1. Technical Field of the Invention
This invention relates to telecommunication systems and, more particularly, to a method of reducing speech delay in digital transmission systems by implementing a single coder-decoder (codec) for an entire speech path.
2. Description of Related Art
Today, in calls from a first mobile subscriber (MS) to a second MS in a cellular telephone network, two codec translations are performed: (1) a translation from an IS-136-based codec to a G.711 codec in a transcoder unit (TRAU) at the originating mobile switching center (MSC), and (2) a translation from a G.711 codec to an IS-136-based codec at a TRAU in the destination serving MSC. These multiple codec translations introduce speech delay and deterioration of speech quality. In an MS-to-MS call, the end-to-end measured average delay is on the order of 210 to 240 milliseconds.
For calls being routed over an Internet Protocol (IP) network, H.323, H.225, and H.245 are a family of ITU standards for packet-based multimedia (audio and video) communications systems. This family of standards permits calls between PC-based phone terminals (clients) in the IP network (LAN or WAN), or between client terminals and phone terminals in the Public Switched Telephone Network (PSTN) or the Public Land Mobile Network (PLMN) in an integrated network. An H.323 Gateway in the H.323 standard provides an interface between the IP network and the PSTN or PLMN, including the audio codec translation.
For a call between an MS in an IS-136 cellular network and an H.323 client in an IP network, the call setup and voice path must go through the H.323 Gateway. Presently, H.323 version 2 Gateways only support G.711 codecs, along with a few other codecs such as G.728. No IS-136-based codecs are supported by the Gateway. Thus, the audio stream, which is transmitted from the calling MS to the originating MSC over the air interface utilizing an IS-136-based codec, must be translated at the originating MSC to G.711. The IS-136-to-G.711 codec translation takes place in a TRAU so that the audio stream coming out of the originating MSC on the pulse code modulation (PCM) link is G.711-coded. When the G.711 codec is utilized between the Gateway and the client, it requires a relatively large packet to transport the 64 Kbs bandwidth required for the voice stream. This worsens the speech delay and jitter problems on the LAN/WAN segment.
Tandem Free Operation (TFO) is an existing methodology utilized to improve speech quality by using only one codec in an MS-to-MS speech path. TFO is applicable only to MS-to-MS calls, however, and does not support calls from an MS to a H.323 client. With TFO, call setup essentially takes place as it normally does without TFO. After the call has been established between the two MSs, a TFO negotiation session begins using inband signaling to establish the type of codec that should be utilized in the call. Only after the completion of this negotiation session does the call get into native codec speech position. As a result, although TFO may improve speech quality by using only one codec in a speech path, there is still an undesirable waiting period in TFO calls before native codec speech can flow due to the TFO negotiation session.
In order to overcome the disadvantage of existing solutions, it would be advantageous to have a method of reducing speech delay in digital transmission systems by implementing a single codec for an entire speech path. Such a method would be performed during call setup, and would not introduce additional speech delays. The present invention provides such a method.